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Abstract:

A solid-state audio power amplifier providing an instantaneous maximum
output voltage capability in excess of its long term power output
capability, in which the input signal is supplied from analogue or
digital signal processor. The signal processor is arranged to limit the
long term power output of the solid-state amplifier in a non-linear
amplitude and frequency dependant manner

Claims:

1. A solid state power amplifier including means for varying the output
voltage in a frequency dependent manner, whereby the power output can be
varied to provide a similar audible effect to that of a valve power
amplifier of equivalent power rating when connected to a guitar
loudspeaker system, the means for the varying the output voltage being
arranged to allow for the frequency response characteristics of the
loudspeaker systems, and having a non-linear amplitude domain transfer
function such that the maximum power output of the amplifier is limited
to at least one of (a) the rated maximum power output of its associated
power supply source, and (b) the maximum rated power input of the
associated loudspeaker system.

2. A solid state power amplifier according to claim 1 in which the output
can be varied to produce similar effects to a valve amplifier in respect
of one or more of sound pressure level characteristics, amplitude
response distortion, and frequency response distortion.

3. A solid state power amplifier according to claim 1 in which the means
for varying the output voltage comprises a digital signal processor.

4. A solid state audio power amplifier according to claim 1 and having a
non-linear frequency response characteristic corresponding to the
combination of a thermionic valve power amplifier connected to a moving
coil guitar loudspeaker system.

5. A solid state audio power amplifier according to claim 1, further
comprising user adjustable means for setting the low and high frequency
amplitude response relative to the middle frequency amplitude response,
so as to correspond to `Presence` and `Resonance` controls provided on
thermionic valve guitar amplifiers.

6. A solid state audio power amplifier according to claim 1 and further
comprising means for varying the non-linear frequency response
characteristic, so as to allow the recreation of the frequency response
attributes of various different thermionic output valve types.

7. A solid state audio power amplifier according to claim 1 and further
comprising means for varying the non-linear frequency response
characteristic, so as to allow the recreation of the tonal attributes of
various different moving coil loudspeaker types and their associated
loudspeaker enclosures.

8. A solid state audio power amplifier according to claim 1 and further
comprising means for varying the non-linear amplitude system input to
system output transfer function, so as to allow the recreation of the
harmonic distortion attributes of various different thermionic output
valve types.

9. A solid state audio power amplifier according to claim 1 and further
comprising means to control and reduce, the maximum power output level
before the onset of a required degree of amplitude distortion becomes
audible.

10. A solid state power amplifier according to claim 2 including means
for storing sets of digital signal processing parameters and coefficients
in digital memory so as to represent a range of harmonic distortion
attributes of various different thermionic output valve types, the
non-linear frequency response characteristics of various different moving
coil loudspeaker types and their associated loudspeaker enclosures, and
the desirable `Presence` and `Resonance` tonal adjustments.

11. A solid state amplifier according to claim 2, including means for
storing sets of digital signal processing parameters and coefficients in
digital memory, so as to represent a range of harmonic distortion
attributes of various different thermionic output valve types, the
non-linear frequency response characteristics of various different moving
coil loudspeaker types and their associated loudspeaker enclosures, and
the desirable `Presence` and `Resonance` tonal adjustments; and further
comprising means for recalling sets of stored digital signal processing
parameters and coefficients to enable the selection of any desired
combination of harmonic distortion attributes of various different
thermionic output valve types.

12. A solid state amplifier according to claim 2, including means for
storing sets of digital signal processing parameters and coefficients in
digital memory, so as to represent a range of harmonic distortion
attributes of various different thermionic output valve types, the
non-linear frequency response characteristics of various different moving
coil loudspeaker types and their associated loudspeaker enclosures, and
the desirable `Presence` and `Resonance` tonal adjustments, and further
comprising means for recalling sets of stored digital signal processing
parameters and coefficients to enable the selection of any desired
combination of harmonic distortion attributes of various different
thermionic output valve types, in which the means for recalling the
digital signal processing parameters and coefficients comprises a user
adjustable hardware control selector connected to digital memory means
whereby the attributes of the different thermionic output valves can be
selected.

13. A solid state power amplifier according to claim 3 in which the
digital signal processor includes IIR filters.

Description:

[0001] This invention relates to the combination of a solid state audio
power amplifier and signal processing means for use with an electric
guitar amplifier.

[0002] It is well known and accepted by the practising electric guitarist,
that a guitar amplifier using thermionic valves (also referred to as
`tubes`) as the primary power amplification devices will be perceived by
the user to sound significantly louder than a guitar amplifier of an
equivalent power output rating utilising solid state power amplification
devices. Additionally, a valve power amplifier will possess desirable
frequency response variations, and, when driven to full power output,
will produce non-linear amplitude and frequency domain distortions that
are also deemed desirable by the practising musician and listener, and
which are not produced by current state of the art solid state linear
audio power amplifiers.

[0003] To overcome the perceived lack of volume, and the lack of both the
desirable frequency response characteristics and the desirable amplitude
distortion characteristics provided by a valve audio power amplifier when
compared to a conventional solid-state power amplifier of the same
nominal power rating, one aspect of this invention provides a combination
of signal processing means and a solid state audio power amplifier and
associated power supply, whose maximum output voltage before limiting is
controlled in a frequency dependant manner such that the maximum RMS
power delivered to an associated guitar loudspeaker system, is equivalent
to that of a conventional valve power amplifier of an equivalent RMS
power rating.

[0004] It is well known and accepted by the users of guitar amplifiers
that utilise thermionic valves (also known as `Tubes`) as the means to
obtain audio power amplification, that a valve amplifier will produce a
higher sound pressure level when used in conjunction with a guitar
loudspeaker system than a solid state (transistorised) audio amplifier of
an equivalent nominal power output rating.

[0005] Over time, various explanations have been suggested for this
phenomenon, all tending to be based around the vague notional concept of
psycho-acoustics. It has been suggested that the inherent non-linearity
in the electrical input/output transfer characteristic of a thermionic
valve, and the resultant addition of harmonically related distortion
components to the output signal that are not present in the original
input signal, has the effect of allowing the user and/or listener of the
valve amplifier, when used in conjunction with a guitar loudspeaker
system, to perceive the sound pressure level of such an amplifier to be
greater than it is in reality.

[0006] This is not the case, and the fundamental cause for the increased
sound pressure level of the system can be shown by a straightforward
engineering analysis of a conventional valve power amplifier driving a
typical musical instrument type loudspeaker system.

[0007] Some embodiments of the invention will now be described by way of
example with reference to the accompanying drawings in which:

[0013]FIG. 6:--Loudspeaker system electrical equivalent model schematic
driven from a audio power amplifier with voltage gain `Aol` and output
resistance `Rout`, with negative feedback factor `Afb` applied to the
power amplifier.

[0015] FIG. 8:--Loudspeaker system electrical model equivalent schematic
driven from a audio power amplifier with voltage gain `Aol` and output
resistance `Rout`, with negative feedback applied to the power amplifier
via frequency selective low-pass and high-pass `PRESENCE` and `RESONANCE`
controls in the negative feedback loop.

[0016]FIG. 9:--Loudspeaker system electrical equivalent model schematic
terminal voltage frequency response plot of system in FIG. 8, for various
settings of the `PRESENCE` control.

[0017]FIG. 10:--Loudspeaker system electrical model equivalent schematic
terminal voltage frequency response plot of system in FIG. 8, for various
settings of the `RESONANCE` control.

[0018]FIG. 11:--Shows the general arrangement of a digital signal
processing unit according to the invention, arranged to receive and
process an audio input signal, with the processed signal output connected
to a audio power amplification stage, in turn driving a loudspeaker.

[0019]FIG. 12:--Depicts in greater detail the digital signal processing
unit of FIG. 11, with analogue to digital conversion means to receive an
audio input signal and digital to analogue conversion means to output an
audio signal, to and from respectively, the digital signal processing
unit. Also illustrated is digital memory means for the storage of audio
data, filter coefficients and program code, as required by the digital
signal processing unit.

[0026] Rvc represents the electrical resistance of the loudspeaker voice
coil, and Lvc represents the inductance of the voice coil formed by
winding the voice coil around the loudspeaker iron pole-piece. Lcom and
Cmas represent respectively the compliance and mass of the loudspeaker
cone and the air load enclosed inside the loudspeaker enclosure, whilst
RIos represents the combined losses of both the mechanical loudspeaker
system and the air enclosed inside the loudspeaker cabinet.

[0027] Using the electrical circuit equivalent of a guitar loudspeaker
enclosure system, the terminal impedance of the driver and enclosure
system can be plotted as a function of frequency, as shown in FIG. 2.
Although loudspeaker drive units and systems are quoted by convention to
have a nominal impedance value (typically 4, 8 or 16 Ohms), it can be
seen from reference to FIG. 2, that the system impedance varies by a
large degree dependant on the frequency of the excitation signal being
applied to the system, with a resonant peak in the lower frequency region
due to the mechanical system resonance formed by the loudspeaker drive
unit and the air load inside the loudspeaker enclosure. The rise in
system impedance at higher frequencies is due to the inductive nature of
the loudspeaker drive unit voice coil. It can be further noted from FIG.
2, that the ratio of the lowest to the highest system impedance through
the audio frequency range is typically in excess of 10:1.

[0028] Referring again to FIG. 1, Voltage source V1 is assumed by
convention to have negligible or zero source impedance, such as is the
case with contemporary solid state audio power amplifier design, and it
is therefore apparent that the voltage across the loudspeaker system
terminals will be independent of the frequency of the signal applied to
the loudspeaker voice coil terminals. This is depicted in FIG. 3.

[0029] Now consider the case where the loudspeaker system is being driven
from an amplifier with a voltage gain of `Aol` and with an intrinsic,
non-zero, output resistance `Rout`, as depicted in FIG. 4. It can be seen
by inspection that the combination of the loudspeaker system impedance
and the amplifier output resistance form a potential divider across the
amplifier output terminals, with Rout forming the upper element of the
potential divider, and the loudspeaker electrical system constituting the
lower element of the potential divider. Due to the frequency dependant
magnitude of the impedances of the various loudspeaker electrical
equivalent circuit elements, the voltage across the loudspeaker system
terminals now becomes highly dependent upon the frequency of the signal
being applied to the combined amplifier and loudspeaker system, and this
voltage will vary according to the frequency of the signal applied to the
input of the combined amplifier and loudspeaker system. This is
illustrated in FIG. 5, which shows the loudspeaker terminal voltage for a
typical Celestion G12-75 twelve inch guitar loudspeaker drive unit
mounted in a sealed enclosure of 40 Litres, when driven from a valve
audio power amplifier typical source impedance of 100 ohms. It can be
observed that there is a variation in excess of twenty decibels in the
voltage developed across the loudspeaker system terminals through the
range of the audio spectrum. Under normal linear loudspeaker drive unit
operating conditions, the sound pressure level produced by a moving coil
loudspeaker system is directly proportional to the terminal voltage
applied to the loudspeaker, and there will therefore be a corresponding
variation in the sound pressure level produced by the loudspeaker system.
Such variation in sound pressure level contrasts markedly with the case
of the system response depicted in FIG. 3.

[0030] Valve audio power amplifiers almost invariably utilise pentode
(five electrode) or tetrode (four electrode) devices as the active power
amplification devices. Typical examples of audio power pentodes are types
EL34 and EL84, with types KT88, 6550 and 6L6 being examples of typical
beam tetrodes. Both types of device are characterised by a transfer
function that closely approximates that of a voltage controlled current
source, and by direct implication, this infers a characteristic high
output resistance to the device. By ensuring that the maximum output
current capability of the particular valve type utilised in a power
amplifier is not exceeded, the maximum output voltage capability of a
valve amplifier is then set by the value of the load resistance that the
valve amplifier is connected to. Referring again to FIG. 4, it can be
seen that the maximum voltage applied by a valve amplifier to typical
loudspeaker system will be highest at the fundamental low frequency
resonance of the loudspeaker and at high frequencies where the inductance
of the voice coil forms a significant part of the total magnitude of the
loudspeaker load impedance.

[0031] This characteristic rise in maximum, undistorted, peak voltage
delivery capability of a valve power amplifier, at the frequency
dependant higher values of the loudspeaker system characteristic
impedance is the fundamental reason that a valve audio power amplifier
will sound louder than a conventional solid state power amplifier of an
equivalent stated nominal power rating.

[0032] It s also noted that over the range of frequencies where the
magnitude of the loudspeaker system impedance rises above the nominal
impedance of the loudspeaker system, the power delivered by the
amplifier, and dissipated in the loudspeaker load, falls. As a direct
consequence, the power input requirement to the amplifier, as supplied by
the power supply unit, will also fall.

[0033] The use of negative feedback in audio power amplifiers, both valve
and solid state, is well known and brings many conventional advantages,
including the reduction of harmonic distortion, increased bandwidth, and
a lowering in system output impedance. All these advantages are
conventionally deemed to be desirable, and it is understood that these
advantages are obtained at the expense of total system closed loop gain.

[0034]FIG. 6 depicts the same arrangement as in FIG. 4, but with the
addition of a negative feedback path, provided by subtracting a fraction
of the output signal generated by the system, Afb, from the input signal
applied to the system. FIG. 7 shows the resultant loudspeaker terminal
frequency response of the combined power amplifier, loudspeaker
electrical load and feedback system. It can be immediately observed from
the frequency response curve that the variation in amplitude response
across the audio frequency range is much reduced as a consequence of the
application of the negative feedback signal.

[0035] By introducing frequency selective filtering into the negative
feedback path of an amplifier, the amplitude response of the amplifier
can be made to be frequency dependant. In 1954 Leo Fender(Fender Musical
Instruments) introduced a variable cut-off frequency low-pass filter into
the feedback path of a valve power amplifier, with the amount of feedback
and the frequency at which the low-pass filtering is introduced being
adjustable via a front panel control, which he termed `Presence`. Many
other amplifier designs subsequently copied this feature, and later a
similar control named `Resonance`, to allow control of the low frequency
response of a power amplifier by high-pass filtering of the power
amplifier feed-back signal, was introduced on many guitar amplifier
designs.

[0036] FIG. 8 depicts the general arrangement of FIG. 6, but with the
inclusion of the frequency selective feedback low-pass and high-pass
filtering arrangements just described. Capacitor `Cres` and the user
adjustable control potentiometer `RESONANCE` perform the high-pass
filtering function, whilst capacitor `Cpres` and the user adjustable
control potentiometer `PRESENCE` form the low-pass filtering function.

[0037]FIG. 9 denotes how the high frequency amplitude response of the
combined power amplifier, the associated loudspeaker load, and the
negative feedback network varies as the Presence control is rotated from
minimum to maximum. (Note that by historical convention, control settings
on guitar amplifiers are almost invariably denoted on a `0` (minimum) to
`10` (maximum) scale, and this convention is followed in this document.)

[0038]FIG. 10 denotes how the low frequency amplitude response of the
combined power amplifier, the associated loudspeaker load, and the
negative feedback network varies as the Resonance control is rotated from
minimum to maximum.

[0039] It is to be noted that not all valve guitar amplifier designs
utilise the application negative feedback around an audio power amplifier
section. The omission of negative feedback in the power amplifier
section, and the resultant significant effect on the frequency response
when such a power amplifier is connected to a guitar loudspeaker system
is deemed desirable by many guitarists. One example of a valve guitar
amplifier with no power amplifier stage negative feedback is the VOX
AC30, first produced in 1957, and still in production currently.

[0040]FIG. 11 depicts a block diagram representation of one embodiment of
the present invention, comprising of an audio signal processing means to
receive and process the applied audio input signal, an audio power
amplifier to enable power amplification of the processed input signal,
and a loudspeaker means to convert the processed and power amplified
input into an acoustic output. Also depicted is a source of power to
provide electrical power to the audio power amplifier and to the signal
processing means.

[0041] The signal processing means may be chosen to receive an audio input
in analogue form, or in a digital representation of the audio input
signal, or in both forms. The form of the applied input signal is not
germane to the invention.

[0042] Similarly, the signal processing means may be chosen to output an
analogue signal corresponding to the processed input signal, or may
output a digital representation of the processed signal input signal. The
form of the processed output signal is not germane to the invention.

[0043] The audio power amplifier may be chosen to be a conventional
analogue audio power amplifier, or power amplification may be achieved
and implemented by pulse width modulation, duty cycle control of a high
frequency carrier signal in order to improve the system power conversion
efficiency. The exact means by which audio power amplification is
achieved is not germane to this invention.

[0044] The power supply source for both the signal processing means, and
the audio power amplifier may be derived by a conventional power line
frequency laminated mains transformer, rectifier and bulk energy storage
capacitors, or alternatively may implemented by high frequency switched
mode techniques, offering higher power conversion efficiency, and weight
and size reduction, at the expense of circuit complexity. Again, the
exact nature of the means of power supply implementation is not germane
to this invention.

[0045]FIG. 12 shows one means of signal processing means, whereby the
input signal is received in analogue form, and converted to a digital
representation of the applied analogue signal by an analogue to digital
convertor. The digital representation of the input signal is then routed
to a Digital Signal Processor (DSP) which acts on the numerical
representation of the applied audio input, with the digitally processed
signal then converted back to an analogue representation of the processed
output signal for application to a subsequent power amplification means.

[0046] Also depicted in FIG. 12, is a means of digital data memory for the
storage of the digital filter coefficients required for the processing of
the received digital audio input signal, and the digital program
instruction codes to define the processing structure. Additionally, the
digital memory also allows the storage of signal processing coefficients
in sets (commonly referred to as `patches`), so as to allow the user the
provision of instantaneous selection and recall of a number of
individually user defined and programmed frequency responses and
amplitude distortion characteristics as may be required.

[0047]FIG. 17 illustrates a front panel selector which is connected to
the digital memory so as to allow the user to select the sets of
parameters and coefficients corresponding to various different output
valves.

[0048]FIG. 13 illustrates in schematic format, the numerical signal flow
implementation of a digital signal filter. Such a filter structure is
known as an `Infinite Impulse Response` (IIR) filter. In this example the
structure of a second-order filter is shown. The filter structure
comprises of four delay elements (denoted by the Z-1 specifier),
each providing a time delay equal to the input signal sampling period.
Such sample period delay elements are easily implemented by the use of
temporary digital data memory within the Digital Signal Processor. Two
delay elements act on the input signal data path of the filter, and two
delay elements act on the output signal data path of the filter. By
suitable numerical scaling (or `weighting`) and summation of the input,
delayed input and delayed output filter element terms, a digital filter
can be implemented by DSP hardware and software that approximates to the
response of any and all of the analogue filter functions shown in the
amplifier and loudspeaker system equivalent schematic shown in FIG. 8.
Current state of the art analogue to digital conversion, signal
processing and digital to analogue conversion accuracy and speed is such
that the deviation in amplitude and frequency response characteristics
between the digital filter and the analogue filter characteristics which
the digital filter has been designed to replicate are not audibly
discernable.

[0049] The filter structure illustrated in FIG. 13 is for the
implementation of a second order characteristic. By means of the same
general arrangement of delay elements, coefficient multipliers and
summations, first, second and higher order filters may be implemented on
suitable signal processing hardware. Due to the finite precision by which
filter coefficients can be represented by digital means, there exists an
upper limit as to the order of filter that may be implemented. For the
filters depicted in FIG. 8, the use of a floating point digital signal
processor, with its inherent large numerical dynamic range, is easily
capable of achieving the required numerical accuracy. One example of such
a device is the "SHARC" (Analog Devices Inc., Norwood, Mass. USA).

[0050] The techniques for the analysis of analogue filter circuitry and
topologies, the conversion of the analogue continuous time domain circuit
behaviour to a discrete time domain sampled numerical implementation, and
the verification and error analysis of the conversion process between
continuous and discrete time systems can be easily and efficiently
performed by Computer Aided Design (CAD) tools such as MATLAB and
SIMULINK (Mathworks Inc.). Such CAD tools can also directly generate the
filter coefficients required for the numerical algorithmic processing
implementation of the desired filter responses.

[0052] The absolute value of the applied system input is obtained using
the identity:

For INPUT≧0, OUTPUT=INPUT

For INPUT<0, OUTPUT=-(INPUT)

[0053] By scaling the output of the absolute value process by a factor of
one-half using multiplier `X2`, and subtracting the resultant scaled
absolute value representation of the input signal from the maximum
permissible full scale range of the system, a gain control coefficient is
produced whose output will vary linearly from the system full scale
positive value when no signal is applied to the input, to one half of the
system full scale positive value when the applied input signal is of a
magnitude positive or negative full scale.

[0054] By further multiplying the original input signal to the non-linear
function generator with the signal resulting from the absolute value,
scale and offset process just described, using multiplier `X1`, a system
input to system output signal transfer function is obtained of the form:

Vout=Vin+(Vin2/2) for -1>Vin<0

Vout=0 for Vin=0

Vout=Vin-(Vin2/2) for 0>Vin<1

[0055] For reasons of clarity of explanation, assume the maximum positive
input magnitude of the signal Vin applied to this system is bounded to
+1, and the maximum negative input magnitude of the signal Vin applied to
this system is bounded to -1. This would be the case implicitly for a
fixed point, fractional signal processor. For a floating point processing
system, the full scale positive and full scale negative input is
constrained to the same fractional range by suitable choice of system
input and system output scaling.

[0056] With the maximum input signal range assumed bounded to
-1<Vin<1, the resultant output signal range defined by the
processing equalities stated above lies between
-0.5≦Vin≦0.5. This implies a system gain of one-half.
However, the instantaneous gain of the system is dependent on the
instantaneous magnitude of the applied input signal, with maximum system
gain occurring at Vin=0. By multiplying the output of the multiplier X1
by a factor of two, the large signal, peak numerical gain of the system
is restored to unity, with system gain increasing toward a value of two
as the applied input signal magnitude tends towards zero. The dependence
of the system incremental gain, and thus the system output, upon the
magnitude of the applied input signal directly imparts an inherent
non-linearity to the amplitude domain system input/output signal
response.

[0057] FIG. 15 plots the system input-to-output transfer characteristic of
the non-linear system just described, from minus full scale input (-1) to
plus full scale (+1) input. From inspection of the plotted transfer
characteristic, it is apparent that the incremental gain of the transfer
function, as illustrated by the gradient of the transfer function curve,
is not constant at any point throughout the bounded range of the applied
input signal. As a direct consequence of this property, amplitude
distortion of the applied input signal occurs in the output signal
generated by the system, producing harmonic distortion components in the
output signal spectrum that were not present in the original input system
signal.

[0058] FIG. 16 plots the output amplitude response of the system depicted
in FIG. 14 when excited by a sinusoidal input source. It can be
immediately observed that at all points, apart from the specific cases
where the applied input signal magnitude is zero or +/- full scale, the
resultant system output magnitude is greater in magnitude than the
applied input signal.

[0059] By sequentially connecting together a number, of non-linear,
harmonic distortion generating processing blocks as described by FIG. 14,
with the output of one block feeding the input to the subsequent
non-linear processing block, the amount of instantaneous signal
compression and thus the amount of harmonic distortion produced at the
system process output can be defined. Each non-linear processing block
adds ˜+6 dB of small-signal gain to the total system, whilst the
total system large signal gain for peak input signal magnitude remains at
unity. By this means the input sensitivity for a given level of total
harmonic distortion can be determined and programmed by the number of
serial iterations of the process described by FIG. 14 that are performed.
Various sensitivity settings for prescribed levels of total harmonic
distortion can be stored in the digital memory depicted in FIG. 12, for
recall by the user as required. If control of distortion sensitivity in
finer resolution increments than the inherent ˜6 dB resolution
obtained by the addition of each discrete distortion processing block as
described is required, a linear, numerical attenuation of between 0 to -6
dB can be inserted at the input to the first non-linear processing block.

[0060] Referring again to FIG. 2, it is seen that the terminal impedance
of a moving coil loudspeaker increases substantially around the region of
the fundamental system resonance of the combined loudspeaker drive unit
and the air enclosed within the loudspeaker enclosure.

[0061] As the system impedance rises at this fundamental system resonance,
the current drawn from the audio power amplifier connected to the
loudspeaker, and thus the power delivery requirement from the power
supply that supplies electrical energy to the said audio power amplifier
also reduces, and thus the power dissipated by the loudspeaker drive unit
also reduces.

[0062] Similarly, as the terminal impedance of a moving coil loudspeaker
system rises at higher frequencies, due to the inductive nature of the
voice coil, the power consumed by the loudspeaker system, and thus the
power required to be delivered by the audio power amplifier and
associated power supply source similarly reduces.

[0063] This invention exploits this fundamental reduction in the input
energy requirement of a moving coil loudspeaker for a given acoustic
power output, in the frequency range centred around the fundamental
system resonance, and at the higher frequencies where the voice coil
inductance becomes a significant part of the total loudspeaker system
impedance, by increasing the maximum voltage applied to the moving coil
system in a frequency selective and amplitude limited manner.

[0064] By increasing the applied terminal voltage to the loudspeaker by
these frequency selective, amplitude limited means, the combined system
acoustic sound pressure level may be raised, in a manner that accurately
models the performance of the same loudspeaker system when connected to a
traditional thermionic valve guitar amplifier.

[0065] By limiting the applied system signal input, by means of the
amplitude domain, non-linear, harmonic generating means depicted in FIG.
14, and subsequently filtering the resultant limited output by means of
cascaded digital IIR filter structures, each having a general structure
as shown in FIG. 13, the amplitude response and frequency response of a
particular combination of output valve type, loudspeaker and associated
loudspeaker enclosure, may be accurately reproduced.